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-rw-r--r--src/inputPlugins/_flac_common.c2
-rw-r--r--src/inputPlugins/aac_plugin.c46
-rw-r--r--src/inputPlugins/audiofile_plugin.c8
-rw-r--r--src/inputPlugins/flac_plugin.c4
-rw-r--r--src/inputPlugins/mod_plugin.c4
-rw-r--r--src/inputPlugins/mp3_plugin.c2
-rw-r--r--src/inputPlugins/mp4_plugin.c8
-rw-r--r--src/inputPlugins/mpc_plugin.c12
-rw-r--r--src/inputPlugins/oggflac_plugin.c4
-rw-r--r--src/inputPlugins/oggvorbis_plugin.c4
-rw-r--r--src/inputPlugins/wavpack_plugin.c9
11 files changed, 51 insertions, 52 deletions
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 22d8774a3..f24e20531 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -162,7 +162,7 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
switch (block->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
data->audio_format.bits = (int8_t)si->bits_per_sample;
- data->audio_format.sampleRate = si->sample_rate;
+ data->audio_format.sample_rate = si->sample_rate;
data->audio_format.channels = (int8_t)si->channels;
data->total_time = ((float)si->total_samples) / (si->sample_rate);
break;
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index a96623e1b..e9b2d7476 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -148,7 +148,7 @@ static size_t adts_find_frame(AacBuffer * b)
static void adtsParse(AacBuffer * b, float *length)
{
unsigned int frames, frameLength;
- int sampleRate = 0;
+ int sample_rate = 0;
float framesPerSec;
/* Read all frames to ensure correct time and bitrate */
@@ -158,9 +158,9 @@ static void adtsParse(AacBuffer * b, float *length)
frameLength = adts_find_frame(b);
if (frameLength > 0) {
if (frames == 0) {
- sampleRate = adtsSampleRates[(b->
- buffer[2] & 0x3c)
- >> 2];
+ sample_rate = adtsSampleRates[(b->
+ buffer[2] & 0x3c)
+ >> 2];
}
if (frameLength > b->bytesIntoBuffer)
@@ -171,7 +171,7 @@ static void adtsParse(AacBuffer * b, float *length)
break;
}
- framesPerSec = (float)sampleRate / 1024.0;
+ framesPerSec = (float)sample_rate / 1024.0;
if (framesPerSec != 0)
*length = (float)frames / framesPerSec;
}
@@ -253,7 +253,7 @@ static float getAacFloatTotalTime(char *file)
float length;
faacDecHandle decoder;
faacDecConfigurationPtr config;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
InputStream inStream;
long bread;
@@ -274,11 +274,11 @@ static float getAacFloatTotalTime(char *file)
fillAacBuffer(&b);
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
- &sampleRate, &channels);
+ &sample_rate, &channels);
#else
- bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
- if (bread >= 0 && sampleRate > 0 && channels > 0)
+ if (bread >= 0 && sample_rate > 0 && channels > 0)
length = 0;
faacDecClose(decoder);
@@ -312,7 +312,7 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
faacDecConfigurationPtr config;
long bread;
struct audio_format audio_format;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
unsigned int sampleCount;
char *sampleBuffer;
@@ -346,9 +346,9 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
- &sampleRate, &channels);
+ &sample_rate, &channels);
#else
- bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
if (bread < 0) {
ERROR("Error not a AAC stream.\n");
@@ -386,12 +386,12 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
- sampleRate = frameInfo.samplerate;
+ sample_rate = frameInfo.samplerate;
#endif
if (!initialized) {
audio_format.channels = frameInfo.channels;
- audio_format.sampleRate = sampleRate;
+ audio_format.sample_rate = sample_rate;
decoder_initialized(mpd_decoder, &audio_format, totalTime);
initialized = 1;
}
@@ -402,11 +402,11 @@ static int aac_stream_decode(struct decoder * mpd_decoder,
if (sampleCount > 0) {
bitRate = frameInfo.bytesconsumed * 8.0 *
- frameInfo.channels * sampleRate /
+ frameInfo.channels * sample_rate /
frameInfo.samples / 1000 + 0.5;
file_time +=
(float)(frameInfo.samples) / frameInfo.channels /
- sampleRate;
+ sample_rate;
}
sampleBufferLen = sampleCount * 2;
@@ -446,7 +446,7 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
faacDecConfigurationPtr config;
long bread;
struct audio_format audio_format;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
unsigned int sampleCount;
char *sampleBuffer;
@@ -484,9 +484,9 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
#ifdef HAVE_FAAD_BUFLEN_FUNCS
bread = faacDecInit(decoder, b.buffer, b.bytesIntoBuffer,
- &sampleRate, &channels);
+ &sample_rate, &channels);
#else
- bread = faacDecInit(decoder, b.buffer, &sampleRate, &channels);
+ bread = faacDecInit(decoder, b.buffer, &sample_rate, &channels);
#endif
if (bread < 0) {
ERROR("Error not a AAC stream.\n");
@@ -522,12 +522,12 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
- sampleRate = frameInfo.samplerate;
+ sample_rate = frameInfo.samplerate;
#endif
if (!initialized) {
audio_format.channels = frameInfo.channels;
- audio_format.sampleRate = sampleRate;
+ audio_format.sample_rate = sample_rate;
decoder_initialized(mpd_decoder, &audio_format,
totalTime);
initialized = 1;
@@ -539,11 +539,11 @@ static int aac_decode(struct decoder * mpd_decoder, char *path)
if (sampleCount > 0) {
bitRate = frameInfo.bytesconsumed * 8.0 *
- frameInfo.channels * sampleRate /
+ frameInfo.channels * sample_rate /
frameInfo.samples / 1000 + 0.5;
file_time +=
(float)(frameInfo.samples) / frameInfo.channels /
- sampleRate;
+ sample_rate;
}
sampleBufferLen = sampleCount * 2;
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 421cdf354..4c08074c4 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -71,14 +71,14 @@ static int audiofile_decode(struct decoder * decoder, char *path)
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
audio_format.bits = (uint8_t)bits;
- audio_format.sampleRate =
+ audio_format.sample_rate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
audio_format.channels =
(uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
- total_time = ((float)frame_count / (float)audio_format.sampleRate);
+ total_time = ((float)frame_count / (float)audio_format.sample_rate);
bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
@@ -97,7 +97,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
decoder_clear(decoder);
current = decoder_seek_where(decoder) *
- audio_format.sampleRate;
+ audio_format.sample_rate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
decoder_command_finished(decoder);
}
@@ -110,7 +110,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
current += ret;
decoder_data(decoder, NULL, 1,
chunk, ret * fs,
- (float)current / (float)audio_format.sampleRate,
+ (float)current / (float)audio_format.sample_rate,
bitRate, NULL);
} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
diff --git a/src/inputPlugins/flac_plugin.c b/src/inputPlugins/flac_plugin.c
index cd8a8efd3..2f3ec88d9 100644
--- a/src/inputPlugins/flac_plugin.c
+++ b/src/inputPlugins/flac_plugin.c
@@ -350,11 +350,11 @@ static int flac_decode_internal(struct decoder * decoder,
break;
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
FLAC__uint64 sampleToSeek = decoder_seek_where(decoder) *
- data.audio_format.sampleRate + 0.5;
+ data.audio_format.sample_rate + 0.5;
if (flac_seek_absolute(flacDec, sampleToSeek)) {
decoder_clear(decoder);
data.time = ((float)sampleToSeek) /
- data.audio_format.sampleRate;
+ data.audio_format.sample_rate;
data.position = 0;
decoder_command_finished(decoder);
} else
diff --git a/src/inputPlugins/mod_plugin.c b/src/inputPlugins/mod_plugin.c
index 9ae9cef16..98bd67f0f 100644
--- a/src/inputPlugins/mod_plugin.c
+++ b/src/inputPlugins/mod_plugin.c
@@ -186,12 +186,12 @@ static int mod_decode(struct decoder * decoder, char *path)
}
audio_format.bits = 16;
- audio_format.sampleRate = 44100;
+ audio_format.sample_rate = 44100;
audio_format.channels = 2;
secPerByte =
1.0 / ((audio_format.bits * audio_format.channels / 8.0) *
- (float)audio_format.sampleRate);
+ (float)audio_format.sample_rate);
decoder_initialized(decoder, &audio_format, 0);
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 60e09a1bb..2990de1ac 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -1030,7 +1030,7 @@ static void initAudioFormatFromMp3DecodeData(mp3DecodeData * data,
struct audio_format * af)
{
af->bits = 16;
- af->sampleRate = (data->frame).header.samplerate;
+ af->sample_rate = (data->frame).header.samplerate;
af->channels = MAD_NCHANNELS(&(data->frame).header);
}
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 6a2d167b2..d284313d4 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -92,7 +92,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
struct audio_format audio_format;
unsigned char *mp4Buffer;
unsigned int mp4BufferSize;
- uint32_t sampleRate;
+ uint32_t sample_rate;
unsigned char channels;
long sampleId;
long numSamples;
@@ -149,7 +149,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
mp4ff_get_decoder_config(mp4fh, track, &mp4Buffer, &mp4BufferSize);
if (faacDecInit2
- (decoder, mp4Buffer, mp4BufferSize, &sampleRate, &channels) < 0) {
+ (decoder, mp4Buffer, mp4BufferSize, &sample_rate, &channels) < 0) {
ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
@@ -157,7 +157,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
return -1;
}
- audio_format.sampleRate = sampleRate;
+ audio_format.sample_rate = sample_rate;
audio_format.channels = channels;
file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
scale = mp4ff_time_scale(mp4fh, track);
@@ -255,7 +255,7 @@ static int mp4_decode(struct decoder * mpd_decoder, InputStream * inStream)
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
#endif
- audio_format.sampleRate = scale;
+ audio_format.sample_rate = scale;
audio_format.channels = frameInfo.channels;
decoder_initialized(mpd_decoder, &audio_format,
total_time);
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index ca37333d3..f74dc8ddc 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -154,7 +154,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
audio_format.bits = 16;
audio_format.channels = info.channels;
- audio_format.sampleRate = info.sample_freq;
+ audio_format.sample_rate = info.sample_freq;
replayGainInfo = newReplayGainInfo();
replayGainInfo->albumGain = info.gain_album * 0.01;
@@ -168,7 +168,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
while (!eof) {
if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
samplePos = decoder_seek_where(mpd_decoder) *
- audio_format.sampleRate;
+ audio_format.sample_rate;
if (mpc_decoder_seek_sample(&decoder, samplePos)) {
decoder_clear(mpd_decoder);
s16 = (int16_t *) chunk;
@@ -201,10 +201,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
if (chunkpos >= MPC_CHUNK_SIZE) {
total_time = ((float)samplePos) /
- audio_format.sampleRate;
+ audio_format.sample_rate;
bitRate = vbrUpdateBits *
- audio_format.sampleRate / 1152 / 1000;
+ audio_format.sample_rate / 1152 / 1000;
decoder_data(mpd_decoder, inStream,
inStream->seekable,
@@ -224,10 +224,10 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
if (decoder_get_command(mpd_decoder) != DECODE_COMMAND_STOP &&
chunkpos > 0) {
- total_time = ((float)samplePos) / audio_format.sampleRate;
+ total_time = ((float)samplePos) / audio_format.sample_rate;
bitRate =
- vbrUpdateBits * audio_format.sampleRate / 1152 / 1000;
+ vbrUpdateBits * audio_format.sample_rate / 1152 / 1000;
decoder_data(mpd_decoder, NULL, inStream->seekable,
chunk, chunkpos, total_time, bitRate,
diff --git a/src/inputPlugins/oggflac_plugin.c b/src/inputPlugins/oggflac_plugin.c
index 3a2db5c03..53f233e0c 100644
--- a/src/inputPlugins/oggflac_plugin.c
+++ b/src/inputPlugins/oggflac_plugin.c
@@ -316,12 +316,12 @@ static int oggflac_decode(struct decoder * mpd_decoder, InputStream * inStream)
}
if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_SEEK) {
FLAC__uint64 sampleToSeek = decoder_seek_where(mpd_decoder) *
- data.audio_format.sampleRate + 0.5;
+ data.audio_format.sample_rate + 0.5;
if (OggFLAC__seekable_stream_decoder_seek_absolute
(decoder, sampleToSeek)) {
decoder_clear(mpd_decoder);
data.time = ((float)sampleToSeek) /
- data.audio_format.sampleRate;
+ data.audio_format.sample_rate;
data.position = 0;
decoder_command_finished(mpd_decoder);
} else
diff --git a/src/inputPlugins/oggvorbis_plugin.c b/src/inputPlugins/oggvorbis_plugin.c
index 0e1d523b9..bf2448605 100644
--- a/src/inputPlugins/oggvorbis_plugin.c
+++ b/src/inputPlugins/oggvorbis_plugin.c
@@ -278,7 +278,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
/*printf("new song!\n"); */
vorbis_info *vi = ov_info(&vf, -1);
audio_format.channels = vi->channels;
- audio_format.sampleRate = vi->rate;
+ audio_format.sample_rate = vi->rate;
if (!initialized) {
float total_time = ov_time_total(&vf, -1);
if (total_time < 0)
@@ -311,7 +311,7 @@ static int oggvorbis_decode(struct decoder * decoder, InputStream * inStream)
decoder_data(decoder, inStream,
inStream->seekable,
chunk, chunkpos,
- ov_pcm_tell(&vf) / audio_format.sampleRate,
+ ov_pcm_tell(&vf) / audio_format.sample_rate,
bitRate, replayGainInfo);
chunkpos = 0;
if (decoder_get_command(decoder) == DECODE_COMMAND_STOP)
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index af7c3a2f3..3e99980bd 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -140,7 +140,7 @@ static void wavpack_decode(struct decoder * decoder,
int position, outsamplesize;
int Bps;
- audio_format.sampleRate = WavpackGetSampleRate(wpc);
+ audio_format.sample_rate = WavpackGetSampleRate(wpc);
audio_format.channels = WavpackGetReducedChannels(wpc);
audio_format.bits = WavpackGetBitsPerSample(wpc);
@@ -168,7 +168,7 @@ static void wavpack_decode(struct decoder * decoder,
samplesreq = sizeof(chunk) / (4 * audio_format.channels);
decoder_initialized(decoder, &audio_format,
- (float)allsamples / audio_format.sampleRate);
+ (float)allsamples / audio_format.sample_rate);
position = 0;
@@ -180,7 +180,7 @@ static void wavpack_decode(struct decoder * decoder,
decoder_clear(decoder);
where = decoder_seek_where(decoder) *
- audio_format.sampleRate;
+ audio_format.sample_rate;
if (WavpackSeekSample(wpc, where)) {
position = where;
decoder_command_finished(decoder);
@@ -200,8 +200,7 @@ static void wavpack_decode(struct decoder * decoder,
int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
1000 + 0.5);
position += samplesgot;
- file_time = (float)position /
- audio_format.sampleRate;
+ file_time = (float)position / audio_format.sample_rate;
format_samples(Bps, chunk,
samplesgot * audio_format.channels);
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-04 19:04:18 +0000